[OpenSIPS-Users] OpenSIPS with Loose routing

Bogdan-Andrei Iancu bogdan at voice-system.ro
Wed Aug 25 11:42:45 CEST 2010


Hi Doug,

Please post a SIP capture of the call in order to understand the 
scenario - the capture should be made from opensips server and it should 
include both inbound and outbound traffic (for opensips).

The idea is that I need to check if the record_route was correctly done 
at INVITE time, if the route set was correctly mirrored by b2bua, etc.....

Regards,
Bogdan

Doug wrote:
>   Hi All,
>
> I'm having an issue with loose routing and call setups.
>
> My call flow looks like the following:
>
> 192.168.112.110 (ATA) -----> 192.168.110.1:5060 (OpenSIPS) ------> 
> 192.168.10.1:5080 (Sippy b2bua) -----> 192.168.10.50:5060 (TDM Gateway - 
> Audiocodes)
>
> Now, OpenSIPS and Sippy B2bua are on the same server (just different 
> listening ports). OpenSIPS is configured with mhomed=1 and listening on 
> 2x IP Addresses - namely 192.168.110.1 and 192.168.10.2
>
> Now the call gets setup correctly, until the ACK is passed through.
>
> The ATA sends an ACK to OpenSIPS with record routing, and the RURI is 
> set as sip:192.168.10.1:5080. Now before, if I called loose_route(), the 
> ACK was sent to the 192.168.10.2 IP of opensips, and in this case, it 
> just loops endlessly.
>
> So I decided to put a check above loose_route, to check if the method 
> was an ACK and then to forward it direct to the RURI instead of having 
> the RURI rewritten based on loose_route(). This fixed the problem perfectly.
>
> The issue is now the ATA attempts to do a T.38 switch over, and the same 
> issues as the ACK happens.
>
> Is there a better way for me to be doing this. I have to use Sippy b2bua 
> due to the billing / radius integration it has - should I disable 
> loose_routing altogether?
>
> By the way, the following code block is what was used to fix the ACK issue:
>
>          if (is_method("ACK"))
>          {
>                  route(16);
>          }
>
>          if(loose_route())
>          {
> ...
>
> route 16 simply checks a transaction and then calls t_relay.
>
> Looking forward to the assistance.
>
> Many thanks
> Doug
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro




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